ref: 6a294daa7859eaf0250aa4a77484dd11550e5c5e
dir: /src/i_sdlsound.c/
// Emacs style mode select -*- C++ -*-
//-----------------------------------------------------------------------------
//
// Copyright(C) 1993-1996 Id Software, Inc.
// Copyright(C) 2005 Simon Howard
// Copyright(C) 2008 David Flater
//
// This program is free software; you can redistribute it and/or
// modify it under the terms of the GNU General Public License
// as published by the Free Software Foundation; either version 2
// of the License, or (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
// 02111-1307, USA.
//
// DESCRIPTION:
// System interface for sound.
//
//-----------------------------------------------------------------------------
#include "config.h"
#include <stdio.h>
#include <stdlib.h>
#include <assert.h>
#include "SDL.h"
#include "SDL_mixer.h"
#ifdef HAVE_LIBSAMPLERATE
#include <samplerate.h>
#endif
#include "deh_main.h"
#include "i_system.h"
#include "s_sound.h"
#include "m_argv.h"
#include "w_wad.h"
#include "z_zone.h"
#include "doomdef.h"
#define LOW_PASS_FILTER
#define NUM_CHANNELS 16
static boolean sound_initialised = false;
static Mix_Chunk sound_chunks[NUMSFX];
static int channels_playing[NUM_CHANNELS];
static int mixer_freq;
static Uint16 mixer_format;
static int mixer_channels;
static uint32_t (*ExpandSoundData)(byte *data, int samplerate, int length,
Mix_Chunk *destination) = NULL;
int use_libsamplerate = 0;
// When a sound stops, check if it is still playing. If it is not,
// we can mark the sound data as CACHE to be freed back for other
// means.
static void ReleaseSoundOnChannel(int channel)
{
int i;
int id = channels_playing[channel];
if (!id)
{
return;
}
channels_playing[channel] = sfx_None;
for (i=0; i<NUM_CHANNELS; ++i)
{
// Playing on this channel? if so, don't release.
if (channels_playing[i] == id)
return;
}
// Not used on any channel, and can be safely released
Z_ChangeTag(sound_chunks[id].abuf, PU_CACHE);
}
#ifdef HAVE_LIBSAMPLERATE
// Returns the conversion mode for libsamplerate to use.
static int SRC_ConversionMode(void)
{
switch (use_libsamplerate)
{
// 0 = disabled
default:
case 0:
return -1;
// Ascending numbers give higher quality
case 1:
return SRC_LINEAR;
case 2:
return SRC_ZERO_ORDER_HOLD;
case 3:
return SRC_SINC_FASTEST;
case 4:
return SRC_SINC_MEDIUM_QUALITY;
case 5:
return SRC_SINC_BEST_QUALITY;
}
}
// libsamplerate-based generic sound expansion function for any sample rate
// unsigned 8 bits --> signed 16 bits
// mono --> stereo
// samplerate --> mixer_freq
// Returns number of clipped samples.
// DWF 2008-02-10 with cleanups by Simon Howard.
static uint32_t ExpandSoundData_SRC(byte *data,
int samplerate,
int length,
Mix_Chunk *destination)
{
SRC_DATA src_data;
uint32_t i, abuf_index=0, clipped=0;
int retn;
int16_t *expanded;
src_data.input_frames = length;
src_data.data_in = malloc(length * sizeof(float));
src_data.src_ratio = (double)mixer_freq / samplerate;
// We include some extra space here in case of rounding-up.
src_data.output_frames = src_data.src_ratio * length + (mixer_freq / 4);
src_data.data_out = malloc(src_data.output_frames * sizeof(float));
assert(src_data.data_in != NULL && src_data.data_out != NULL);
// Convert input data to floats
for (i=0; i<length; ++i)
{
// Unclear whether 128 should be interpreted as "zero" or whether a
// symmetrical range should be assumed. The following assumes a
// symmetrical range.
src_data.data_in[i] = data[i] / 127.5 - 1;
}
// Do the sound conversion
retn = src_simple(&src_data, SRC_ConversionMode(), 1);
assert(retn == 0);
// Convert the result back into 16-bit integers.
destination->alen = src_data.output_frames_gen * 4;
destination->abuf = Z_Malloc(destination->alen, PU_STATIC,
&destination->abuf);
expanded = (int16_t *) destination->abuf;
for (i=0; i<src_data.output_frames_gen; ++i)
{
// libsamplerate does not limit itself to the -1.0 .. 1.0 range on
// output, so a multiplier less than INT16_MAX (32767) is required
// to avoid overflows or clipping. However, the smaller the
// multiplier, the quieter the sound effects get, and the more you
// have to turn down the music to keep it in balance.
// 22265 is the largest multiplier that can be used to resample all
// of the Vanilla DOOM sound effects to 48 kHz without clipping
// using SRC_SINC_BEST_QUALITY. It is close enough (only slightly
// too conservative) for SRC_SINC_MEDIUM_QUALITY and
// SRC_SINC_FASTEST. PWADs with interestingly different sound
// effects or target rates other than 48 kHz might still result in
// clipping--I don't know if there's a limit to it.
// As the number of clipped samples increases, the signal is
// gradually overtaken by noise, with the loudest parts going first.
// However, a moderate amount of clipping is often tolerated in the
// quest for the loudest possible sound overall. The results of
// using INT16_MAX as the multiplier are not all that bad, but
// artifacts are noticeable during the loudest parts.
float cvtval_f = src_data.data_out[i] * 22265;
int32_t cvtval_i = cvtval_f + (cvtval_f < 0 ? -0.5 : 0.5);
// Asymmetrical sound worries me, so we won't use -32768.
if (cvtval_i < -INT16_MAX) {
cvtval_i = -INT16_MAX;
++clipped;
} else if (cvtval_i > INT16_MAX) {
cvtval_i = INT16_MAX;
++clipped;
}
// Left and right channels
expanded[abuf_index++] = cvtval_i;
expanded[abuf_index++] = cvtval_i;
}
free(src_data.data_in);
free(src_data.data_out);
return clipped;
}
#endif
static boolean ConvertibleRatio(int freq1, int freq2)
{
int ratio;
if (freq1 > freq2)
{
return ConvertibleRatio(freq2, freq1);
}
else if ((freq2 % freq1) != 0)
{
// Not in a direct ratio
return false;
}
else
{
// Check the ratio is a power of 2
ratio = freq2 / freq1;
while ((ratio & 1) == 0)
{
ratio = ratio >> 1;
}
return ratio == 1;
}
}
// Generic sound expansion function for any sample rate.
// Returns number of clipped samples (always 0).
static uint32_t ExpandSoundData_SDL(byte *data,
int samplerate,
int length,
Mix_Chunk *destination)
{
SDL_AudioCVT convertor;
uint32_t expanded_length;
// Calculate the length of the expanded version of the sample.
expanded_length = (uint32_t) ((((uint64_t) length) * mixer_freq) / samplerate);
// Double up twice: 8 -> 16 bit and mono -> stereo
expanded_length *= 4;
destination->alen = expanded_length;
destination->abuf
= Z_Malloc(expanded_length, PU_STATIC, &destination->abuf);
// If we can, use the standard / optimised SDL conversion routines.
if (samplerate <= mixer_freq
&& ConvertibleRatio(samplerate, mixer_freq)
&& SDL_BuildAudioCVT(&convertor,
AUDIO_U8, 1, samplerate,
mixer_format, mixer_channels, mixer_freq))
{
convertor.buf = destination->abuf;
convertor.len = length;
memcpy(convertor.buf, data, length);
SDL_ConvertAudio(&convertor);
}
else
{
Sint16 *expanded = (Sint16 *) destination->abuf;
int expanded_length;
int expand_ratio;
int i;
// Generic expansion if conversion does not work:
//
// SDL's audio conversion only works for rate conversions that are
// powers of 2; if the two formats are not in a direct power of 2
// ratio, do this naive conversion instead.
// number of samples in the converted sound
expanded_length = ((uint64_t) length * mixer_freq) / samplerate;
expand_ratio = (length << 8) / expanded_length;
for (i=0; i<expanded_length; ++i)
{
Sint16 sample;
int src;
src = (i * expand_ratio) >> 8;
sample = data[src] | (data[src] << 8);
sample -= 32768;
// expand 8->16 bits, mono->stereo
expanded[i * 2] = expanded[i * 2 + 1] = sample;
}
#ifdef LOW_PASS_FILTER
// Perform a low-pass filter on the upscaled sound to filter
// out high-frequency noise from the conversion process.
{
float rc, dt, alpha;
// Low-pass filter for cutoff frequency f:
//
// For sampling rate r, dt = 1 / r
// rc = 1 / 2*pi*f
// alpha = dt / (rc + dt)
// Filter to the half sample rate of the original sound effect
// (maximum frequency, by nyquist)
dt = 1.0f / mixer_freq;
rc = 1.0f / (3.14f * samplerate);
alpha = dt / (rc + dt);
for (i=1; i<expanded_length; ++i)
{
expanded[i] = (Sint16) (alpha * expanded[i] + (1 - alpha) * expanded[i-1]);
}
}
#endif /* #ifdef LOW_PASS_FILTER */
}
return 0;
}
// Load and convert a sound effect
// Returns true if successful
static boolean CacheSFX(int sound)
{
int lumpnum;
unsigned int lumplen;
int samplerate;
int clipped;
unsigned int length;
byte *data;
// need to load the sound
lumpnum = S_sfx[sound].lumpnum;
data = W_CacheLumpNum(lumpnum, PU_STATIC);
lumplen = W_LumpLength(lumpnum);
// Check the header, and ensure this is a valid sound
if (lumplen < 8
|| data[0] != 0x03 || data[1] != 0x00)
{
// Invalid sound
return false;
}
// 16 bit sample rate field, 32 bit length field
samplerate = (data[3] << 8) | data[2];
length = (data[7] << 24) | (data[6] << 16) | (data[5] << 8) | data[4];
// If the header specifies that the length of the sound is greater than
// the length of the lump itself, this is an invalid sound lump
if (length > lumplen - 8)
{
return false;
}
// Sample rate conversion
// DWF 2008-02-10: sound_chunks[sound].alen and abuf are determined
// by ExpandSoundData.
sound_chunks[sound].allocated = 1;
sound_chunks[sound].volume = MIX_MAX_VOLUME;
clipped = ExpandSoundData(data + 8,
samplerate,
length,
&sound_chunks[sound]);
if (clipped)
{
fprintf(stderr, "Sound %d: clipped %u samples (%0.2f %%)\n",
sound, clipped,
400.0 * clipped / sound_chunks[sound].alen);
}
// don't need the original lump any more
W_ReleaseLumpNum(lumpnum);
return true;
}
#ifdef HAVE_LIBSAMPLERATE
// Preload all the sound effects - stops nasty ingame freezes
static void I_PrecacheSounds(void)
{
char namebuf[9];
int i;
printf("I_PrecacheSounds: Precaching all sound effects..");
for (i=sfx_pistol; i<NUMSFX; ++i)
{
if ((i % 6) == 0)
{
printf(".");
fflush(stdout);
}
sprintf(namebuf, "ds%s", DEH_String(S_sfx[i].name));
S_sfx[i].lumpnum = W_CheckNumForName(namebuf);
if (S_sfx[i].lumpnum != -1)
{
CacheSFX(i);
if (sound_chunks[i].abuf != NULL)
{
Z_ChangeTag(sound_chunks[i].abuf, PU_CACHE);
}
}
}
printf("\n");
}
#endif
static Mix_Chunk *GetSFXChunk(int sound_id)
{
if (sound_chunks[sound_id].abuf == NULL)
{
if (!CacheSFX(sound_id))
return NULL;
}
else
{
// don't free the sound while it is playing!
Z_ChangeTag(sound_chunks[sound_id].abuf, PU_STATIC);
}
return &sound_chunks[sound_id];
}
//
// Retrieve the raw data lump index
// for a given SFX name.
//
static int I_SDL_GetSfxLumpNum(sfxinfo_t* sfx)
{
char namebuf[9];
sprintf(namebuf, "ds%s", DEH_String(sfx->name));
return W_GetNumForName(namebuf);
}
static void I_SDL_UpdateSoundParams(int handle, int vol, int sep)
{
int left, right;
if (!sound_initialised)
{
return;
}
left = ((254 - sep) * vol) / 127;
right = ((sep) * vol) / 127;
Mix_SetPanning(handle, left, right);
}
//
// Starting a sound means adding it
// to the current list of active sounds
// in the internal channels.
// As the SFX info struct contains
// e.g. a pointer to the raw data,
// it is ignored.
// As our sound handling does not handle
// priority, it is ignored.
// Pitching (that is, increased speed of playback)
// is set, but currently not used by mixing.
//
static int I_SDL_StartSound(int id, int channel, int vol, int sep)
{
Mix_Chunk *chunk;
if (!sound_initialised)
{
return -1;
}
// Release a sound effect if there is already one playing
// on this channel
ReleaseSoundOnChannel(channel);
// Get the sound data
chunk = GetSFXChunk(id);
if (chunk == NULL)
{
return -1;
}
// play sound
Mix_PlayChannelTimed(channel, chunk, 0, -1);
channels_playing[channel] = id;
// set separation, etc.
I_SDL_UpdateSoundParams(channel, vol, sep);
return channel;
}
static void I_SDL_StopSound (int handle)
{
if (!sound_initialised)
{
return;
}
Mix_HaltChannel(handle);
// Sound data is no longer needed; release the
// sound data being used for this channel
ReleaseSoundOnChannel(handle);
}
static boolean I_SDL_SoundIsPlaying(int handle)
{
if (handle < 0)
{
return false;
}
return Mix_Playing(handle);
}
//
// Periodically called to update the sound system
//
static void I_SDL_UpdateSound(void)
{
int i;
// Check all channels to see if a sound has finished
for (i=0; i<NUM_CHANNELS; ++i)
{
if (channels_playing[i] && !I_SDL_SoundIsPlaying(i))
{
// Sound has finished playing on this channel,
// but sound data has not been released to cache
ReleaseSoundOnChannel(i);
}
}
}
static void I_SDL_ShutdownSound(void)
{
if (!sound_initialised)
{
return;
}
Mix_CloseAudio();
SDL_QuitSubSystem(SDL_INIT_AUDIO);
sound_initialised = false;
}
static boolean I_SDL_InitSound(void)
{
int i;
// No sounds yet
for (i=0; i<NUMSFX; ++i)
{
sound_chunks[i].abuf = NULL;
}
for (i=0; i<NUM_CHANNELS; ++i)
{
channels_playing[i] = sfx_None;
}
if (SDL_Init(SDL_INIT_AUDIO) < 0)
{
fprintf(stderr, "Unable to set up sound.\n");
return false;
}
if (Mix_OpenAudio(snd_samplerate, AUDIO_S16SYS, 2, 1024) < 0)
{
fprintf(stderr, "Error initialising SDL_mixer: %s\n", Mix_GetError());
return false;
}
ExpandSoundData = ExpandSoundData_SDL;
Mix_QuerySpec(&mixer_freq, &mixer_format, &mixer_channels);
#ifdef HAVE_LIBSAMPLERATE
if (use_libsamplerate != 0)
{
if (SRC_ConversionMode() < 0)
{
I_Error("I_SDL_InitSound: Invalid value for use_libsamplerate: %i",
use_libsamplerate);
}
ExpandSoundData = ExpandSoundData_SRC;
I_PrecacheSounds();
}
#else
if (use_libsamplerate != 0)
{
fprintf(stderr, "I_SDL_InitSound: use_libsamplerate=%i, but "
"libsamplerate support not compiled in.\n",
use_libsamplerate);
}
#endif
Mix_AllocateChannels(NUM_CHANNELS);
SDL_PauseAudio(0);
sound_initialised = true;
return true;
}
static snddevice_t sound_sdl_devices[] =
{
SNDDEVICE_SB,
SNDDEVICE_PAS,
SNDDEVICE_GUS,
SNDDEVICE_WAVEBLASTER,
SNDDEVICE_SOUNDCANVAS,
SNDDEVICE_AWE32,
};
sound_module_t sound_sdl_module =
{
sound_sdl_devices,
arrlen(sound_sdl_devices),
I_SDL_InitSound,
I_SDL_ShutdownSound,
I_SDL_GetSfxLumpNum,
I_SDL_UpdateSound,
I_SDL_UpdateSoundParams,
I_SDL_StartSound,
I_SDL_StopSound,
I_SDL_SoundIsPlaying,
};